similar to: Polycom and multicast

Does polycom support "normal" multicast from asterisk as the source? I'm getting the impression that it only supports its OWN phone to phone multicast or something. Thanks, Jerry

I am using 11.17.0 - and MulticastRTP. Doesnt seem to work with polycom phones as other devices receive my multicast just fine. Is there something special to do to get multicast working with polycom phones? (other than enable multicast on the actual phone). Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

> I am using 11.17.0 - and MulticastRTP. Doesnt seem to work with > polycom phones as other devices receive my multicast just fine. > > Is there something special to do to get multicast working with polycom phones? > (other than enable multicast on the actual phone). Didn't see if anyone had answered you or not on this, but Polycom uses their own form of MulticastRTP. It

What is the default multicast codec for multicast in Asterisk 13 ? G.729 or G.711 or other ? Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200401/b83fc8ca/attachment.html>

Hi all, I want to send a text message to a polycom phone. I know how to create a call file - but that will "call" the phone and nothing happens till the phone is answered. How do I create a call file that will "send" a text message over SIP to the polycom phone? So the phone does not have to answer - just shows the message. Thanks, Jerry -------------- next part

I am using asterisk 13.33.0 and POlycom phone with the latest firmware. The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted - it receives a call, after some time (couple minutes) it no longer receives a ring. I see no errors in the CLI - looks just like the previous call as far as I can tell. Then reboot the phone and as soon as its ready call it

I am trying to find out what codec is used in the asterisk multicast ? Is it ulaw, alaw, g.729 or something else ? Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200216/d47e6c66/attachment.html>

On Mon, 13 Apr 2015, Kevin Larsen wrote: > I hesitate to promote the name here since this is non-commercial > discussion... > but Polycom... > Polycom phones... If mentioning Polycom is OK, I think mentioning a possible commercial solution is OK. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at

I'm currently working with Asterisk 11.8.1 trying to get Multicast RTP working (it's not) with some Polycom phones, and I'm really trying to determine if Asterisk or the phones are the issue. I THINK it's Asterisk... In extensions.conf I have a simple: "Page(MulticastRTP/basic/x.x.x.x:xxxx) line, and when I dial that extension I get: -- Called

I was trying to get my polycom phone to auto answer. I added this to the dialplan. Used a different phone to call "22" and the phone rang it did not auto answer. Did I miss something? exten => 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten => 22,n,SipAddHeader(Alert-Info: Ring Answer) exten => 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten =>

Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light myself. Thanks, Jerry

I have a polycom 501 phone that is 1 hour off now. Before last sunday (time change) the time was fine. <?xml version="1.0" standalone="yes"?> <PHONE_CONFIG> <OVERRIDES _.0x20._log.level.change.sip="0" tcpIpApp.sntp.daylightSavings.stop.date="4" tcpIpApp.sntp.daylightSavings.stop.month="11"

I recently got a polycom 501. I was trying to get the phone to accept the TFTP boot files. I was REALLY confused when I finally figured out that the phone does FTP by default and you have to go change it to TFTP using the keyboard menus to switch it to TFTP. Am I missing something here? I certainly would have thought the phone would be intelligent enough to try a FTP first - If you dont get

Well, yes and no. I tested that before and it causes a silent ring instead of a call rejection. I actually want to disable the entire feature. So the phone always rings unless you're actually on the phone. Thanks for the reply though! Regards, Jan ________________________________ Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Jerry

Hi All, I have polycom phones setup in an office connected to a cloud asterisk server. The polycom phones can call out just fine - audio just fine. However a call coming into the cloud asterisk answers fine - get the autoattendant, enter the extension and the polycom does not ring. The CLI shows that the correct SIP extension is being Dialed (SIP/524) Looks like I'm getting a 401 permission

I have a machine with three IP addresses. NIC eth0 NIC eth1 and a virtual address on ETH1 All my devices work normally communicating to the virtual address on eth1. My question is just for mulitcast. The end device has an option for "allowed source" so I put in the virtual address from my server. No multicast audio received... I then disabled the "allowed source" and tried

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Jerry Glomph Black > Sent: Monday, January 30, 2006 11:59 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Polycom IP301: Pass-through ethernet port > unusable? > > Have just done a

I just tried this in my extensions.conf exten => **,1,Noop(Testing) exten => **,n,Playback(demo-congrats) Did a reload... and the above does not happen. I created as 12 instead of the ** and that works fine. Is there anyway to get the ** to work? I also am using a polycom phone if that affects things. I'm using asterisk 13.15.0 Thanks Jerry -------------- next part --------------

Could not find your post for 4 months ago. -------------- Original message -------------- From: "Anton Krall" <akrall-lists@intruder.com.mx> > Yes, check a post that I made about 4 months ago, I posted the cofig for > setting the speaker, handset and ring volumes .. > > |-----Original Message----- > |From: asterisk-users-bounces@lists.digium.com >

Hi all, I upgraded from asterisk 1.2.23 and zaptel 1.2.19 to asterisk 1.4.18 and zaptel 1.4.12.1 I use polycom 501 phones internally. Everything seems fine. I can pick up the phone and call out, calls coming in work just fine. The issue I see is when the system first calls me, then calls someone else. This works if its polycom to polycom. I hear audio full channel. If I do polycom to external